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gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
00d566ec297d52c8e732d7299d69e4e580707d61
/
video_engine
/
vie_channel.cc
0291c80
Removed ViE file API.
by mflodman@webrtc.org
· 11 years ago
20cfda6
Remove unused multi stream bandwidth estimator.
by solenberg@webrtc.org
· 11 years ago
b4c89a4
Making no NACK mode work again in VideoEngine.
by mflodman@webrtc.org
· 11 years ago
695ff2a
Add support for padding in pacer.
by stefan@webrtc.org
· 11 years ago
54b6ebc
Correctly set SSRCs for extra send RTP modules.
by stefan@webrtc.org
· 11 years ago
4e5f983
Fix a return value mismatch introduced in r4129.
by stefan@webrtc.org
· 11 years ago
6696fba
Breaking out RTP header parsing from the RTP module.
by stefan@webrtc.org
· 11 years ago
f40e9b6
- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup.
by solenberg@webrtc.org
· 11 years ago
eef4fd5
Adds integration test for RTX and fixes bugs found.
by stefan@webrtc.org
· 11 years ago
453f9c0
Add functions to ViE API to enable/disable the absolute send time header extension.
by solenberg@webrtc.org
· 11 years ago
281cff8
Include files from webrtc/.. paths in video_engine/
by pbos@webrtc.org
· 11 years ago
ac6d919
Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation.
by andresp@webrtc.org
· 11 years ago
06ad384
Trigger a PLI if the duration of non-decodable frames exceeds a threshold.
by stefan@webrtc.org
· 11 years ago
7bc7e02
Adding a payload type for RTX.
by mflodman@webrtc.org
· 11 years ago
67879bc
WebRtc_Word32 -> int32_t in video_engine/
by pbos@webrtc.org
· 11 years ago
208a648
Always set render delay in ViEChannel::RegisterExternalDecoder.
by pbos@webrtc.org
· 11 years ago
065b64d
Remove UDP transport API from ViE
by pwestin@webrtc.org
· 11 years ago
946d240
Adding RTX on source
by mikhal@webrtc.org
· 11 years ago
912b7f7
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004
by pwestin@webrtc.org
· 11 years ago
b793abe
Increasing size of nack list in buffered mode.
by mikhal@webrtc.org
· 11 years ago
2daec4c
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
by pwestin@webrtc.org
· 11 years ago
78e450f
Enabling bufffering mode with no sync module or VoE
by mikhal@webrtc.org
· 11 years ago
55e6f58
Stop and restart fix.
by mflodman@webrtc.org
· 12 years ago
0329e59
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
by fischman@webrtc.org
· 12 years ago
08998cd
fixing nack list size calculation
by mikhal@webrtc.org
· 12 years ago
9d6fcb3
Adding a receive side API for buffering mode.
by mikhal@webrtc.org
· 12 years ago
0c66de6
Updates to send side streaming mode:
by mikhal@webrtc.org
· 12 years ago
cd1ac8b
Don't report an error for GetEstimatedReceiveBandwidth if there is no valid
by mflodman@webrtc.org
· 12 years ago
1619664
Adding a send side API for streaming
by mikhal@webrtc.org
· 12 years ago
7fff32c
Fix mismatch between different NACK list lengths and packet buffers.
by stefan@webrtc.org
· 12 years ago
2a5dbce
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
by stefan@webrtc.org
· 12 years ago
f314c80
Added API to get receive side video delay.
by mflodman@webrtc.org
· 12 years ago
78696d3
Wire up CallStats to provide modules with correct RTT.
by mflodman@webrtc.org
· 12 years ago
5e87b5f
Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016
by pwestin@webrtc.org
· 12 years ago
07e96da
Reorganize modules/video_render.
by andrew@webrtc.org
· 12 years ago
b6d9cfc
Revert the revert in r2988 since that wasn't the issue.
by mflodman@webrtc.org
· 12 years ago
9f269d2
Reverse Merged r2884 & r2888 from trunk.
by vikasmarwaha@webrtc.org
· 12 years ago
3bbed74
Switching to I420VideoFrame
by mikhal@webrtc.org
· 12 years ago
b015cbe
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago